Legal claims defining the scope of protection. Each claim is shown in both the original legal language and a plain English translation.
1. A method for improving video streaming performance of a video in a system having a client machine and remote machine, the method being performed by the client machine and comprising: determining a first number based on one or more parameters, at least one of the parameters being related to current network conditions; determining a second number corresponding to a number of video segments of the video, as calculated by a total size of the video segments, that is greater than or equal in size to a third number determined based on at least a bandwidth-delay product of the network to the remote machine, the third number being no less than two; requesting from the remote machine the second number of video segments in a pipelined fashion, wherein a subsequent request for a video segment of the video is made before a response to a prior request is at least partially received, provided that no less than the second number of video segments are outstanding at any one time, and wherein another subsequent request is made if fewer than the second number of video segments are outstanding; and stopping subsequent pipelined requests if a predetermined size of the video has been requested that is greater than or equal to the first number.
This invention relates to improving video streaming performance in a system with a client machine and a remote machine. The problem addressed is inefficient video streaming due to network conditions, leading to buffering or delays. The method involves dynamically adjusting the number of video segments requested in a pipelined manner to optimize performance. The client machine determines a first number based on network conditions, such as bandwidth or latency. It also calculates a second number representing the number of video segments to request, ensuring the total size of these segments is at least as large as a third number derived from the network's bandwidth-delay product. The third number is set to no less than two segments. The client then requests the second number of segments in a pipelined fashion, where subsequent requests are made before prior responses are fully received, maintaining at least the second number of outstanding requests. If fewer than the second number of segments are outstanding, another request is issued. The pipelining stops once a predetermined size of the video, equal to or greater than the first number, has been requested. This approach ensures efficient use of network resources while minimizing buffering.
2. The method of claim 1 , wherein the requests occur via HTTP.
This invention relates to a method for processing requests in a networked system, specifically focusing on handling requests transmitted via the HTTP protocol. The method involves receiving requests from one or more clients, where these requests are formatted according to the HTTP standard. The system processes these requests by validating their structure, extracting relevant data, and determining the appropriate response based on predefined rules or logic. The method ensures that the requests adhere to HTTP conventions, such as proper headers, status codes, and payload formats, to maintain compatibility with web-based applications and services. Additionally, the method may include error handling mechanisms to manage malformed or unauthorized requests, ensuring robust and secure communication between clients and servers. The invention aims to improve the efficiency and reliability of HTTP-based interactions in distributed systems, particularly in environments where multiple clients interact with a centralized server or service. By standardizing request processing, the method reduces the likelihood of errors and enhances interoperability across different web technologies and platforms.
3. The method of claim 1 , wherein the current network conditions include an estimated bandwidth to the remote machine.
A system and method for optimizing data transmission between a local machine and a remote machine over a network involves monitoring current network conditions to dynamically adjust data transfer parameters. The method includes determining the estimated bandwidth available to the remote machine, which is a key factor in assessing network performance. By evaluating this bandwidth, the system can adapt transmission rates, packet sizes, or other network protocols to improve efficiency and reliability. The method may also involve analyzing latency, packet loss, or other network metrics to further refine data transfer strategies. The goal is to enhance data delivery performance by tailoring transmission settings based on real-time network conditions, ensuring optimal use of available bandwidth and minimizing disruptions. This approach is particularly useful in environments with fluctuating network performance, such as wireless or long-distance connections, where adaptive adjustments are necessary to maintain stable and efficient communication.
4. The method of claim 1 , wherein the current network conditions include an estimate of the network latency or round-trip time to the remote machine.
A system and method for optimizing data transmission in a networked environment addresses the challenge of inefficient data transfer due to unaccounted network conditions. The invention monitors and evaluates real-time network conditions, including latency and round-trip time (RTT) to a remote machine, to dynamically adjust data transmission parameters. By incorporating latency and RTT estimates, the system improves transmission efficiency, reduces delays, and enhances reliability in data exchange between local and remote devices. The method involves continuously assessing network performance metrics, such as latency, to determine optimal transmission strategies. These strategies may include adjusting packet size, transmission rate, or protocol settings to mitigate the impact of network delays. The system ensures adaptive and responsive data handling, particularly in scenarios where network conditions fluctuate, such as in wireless or long-distance communications. The inclusion of latency and RTT measurements allows for precise adjustments, leading to more predictable and efficient data transfers. This approach is applicable in various networked applications, including cloud computing, remote access, and real-time data streaming, where minimizing latency and optimizing throughput are critical.
5. The method of claim 1 , wherein determining the first number includes determining a first number based on two or more parameters, at least two of the parameters being related to network conditions.
This invention relates to a method for optimizing network performance by dynamically determining a first number based on network conditions. The method involves analyzing multiple parameters related to network conditions, such as latency, bandwidth, packet loss, or signal strength, to calculate the first number, which is used to adjust network operations. The first number may influence routing decisions, resource allocation, or transmission settings to improve efficiency and reliability. By incorporating at least two network-related parameters, the method ensures more accurate and adaptive adjustments compared to systems relying on a single metric. This approach helps mitigate issues like congestion, delays, or poor connectivity, enhancing overall network performance. The method can be applied in various network environments, including wireless, wired, or hybrid systems, to dynamically respond to changing conditions and optimize data transmission. The invention aims to provide a more robust and efficient network management solution by leveraging real-time network data.
6. The method of claim 5 , wherein one of the two parameters is an estimated bandwidth to the remote machine.
This invention relates to optimizing data transfer between a local machine and a remote machine by dynamically adjusting parameters based on network conditions. The problem addressed is inefficient data transfer due to static or suboptimal settings, leading to slow transmission speeds or excessive resource usage. The method involves monitoring network performance metrics and dynamically adjusting two key parameters to improve transfer efficiency. One of these parameters is the estimated bandwidth to the remote machine, which is used to predict available network capacity. The other parameter may include factors such as packet size, transmission rate, or error correction settings. By continuously evaluating network conditions and adjusting these parameters in real-time, the system ensures optimal data transfer performance. The method may also involve comparing current performance against historical data or predefined thresholds to determine the most effective parameter adjustments. This dynamic approach reduces latency and improves throughput, particularly in environments with fluctuating network conditions. The invention is applicable to various data transfer scenarios, including file transfers, streaming, and cloud computing, where network efficiency is critical.
7. The method of claim 5 , wherein one of the two parameters is an estimate of network latency or round-trip time to the remote machine.
This invention relates to network communication systems, specifically methods for optimizing data transmission between a local machine and a remote machine. The problem addressed is the inefficiency in network communication due to unpredictable latency or round-trip time (RTT) between machines, which can degrade performance in applications requiring real-time or low-latency responses. The method involves monitoring and adjusting network communication parameters based on real-time conditions. One of the key parameters used for adjustment is an estimate of network latency or round-trip time to the remote machine. This estimate is derived from measuring the time taken for data packets to travel to the remote machine and receive a response. The method dynamically adjusts transmission settings, such as packet size, transmission rate, or retransmission policies, to optimize performance based on the observed latency or RTT. Additionally, the method may involve comparing the estimated latency or RTT against predefined thresholds to determine whether adjustments are necessary. If the latency exceeds a certain threshold, the system may reduce packet size or increase retransmission intervals to mitigate delays. Conversely, if latency is low, the system may increase packet size or transmission rate to improve throughput. By continuously monitoring and adapting to network conditions, the method ensures efficient and reliable data transmission, particularly in environments where network performance is variable. This approach is applicable to various networked applications, including real-time communication, cloud computing, and distributed systems.
8. The method of claim 1 , wherein determining the first number includes determining the first number based on an underperformance parameter used to determine how close performance should be to an optimal value.
A system and method for optimizing performance in a technical process involves determining a first number representing a performance metric. This first number is calculated based on an underperformance parameter, which defines how close the actual performance should be to an optimal value. The underperformance parameter acts as a threshold or tolerance level, ensuring that the performance metric remains within an acceptable range of the optimal value. The method may also include adjusting operational parameters or control settings to maintain or improve performance based on this determination. The system may monitor real-time data, compare it against the optimal value, and use the underperformance parameter to decide whether adjustments are necessary. This approach helps maintain efficiency, reduce waste, or enhance output quality in industrial, manufacturing, or computational processes where performance deviations from optimal conditions are undesirable. The underperformance parameter can be dynamically adjusted based on changing conditions or user-defined criteria to ensure continuous optimization.
9. The method of claim 1 , wherein determining the first number includes determining the first number based on TCP estimates.
This invention relates to network communication systems, specifically methods for optimizing data transmission by dynamically adjusting transmission parameters based on network conditions. The problem addressed is inefficient data transfer due to inaccurate or outdated network performance estimates, leading to suboptimal throughput, latency, or resource utilization. The method involves determining a first number representing a transmission parameter, such as a congestion window size or packet rate, by analyzing Transmission Control Protocol (TCP) estimates. TCP estimates include metrics like round-trip time (RTT), packet loss rate, or bandwidth-delay product, which are used to infer network conditions. By leveraging these estimates, the method dynamically adjusts the transmission parameter to improve data transfer efficiency. For example, if TCP estimates indicate high packet loss, the method may reduce the transmission rate to avoid further congestion. Conversely, if estimates suggest available bandwidth, the method may increase the transmission rate to maximize throughput. The method may also incorporate additional techniques, such as predictive modeling or machine learning, to refine the transmission parameter based on historical or real-time data. This ensures adaptive behavior in varying network environments, such as wireless, wired, or hybrid networks. The goal is to enhance reliability, reduce latency, and optimize resource usage in data transmission systems.
10. The method of claim 9 , wherein the TCP estimates include a number of round-trips before TCP reaches a slow-start threshold or a bandwidth-delay-product.
This invention relates to network performance optimization, specifically improving Transmission Control Protocol (TCP) congestion control mechanisms. The problem addressed is inefficient bandwidth utilization during the slow-start phase of TCP connections, where the protocol gradually increases transmission rates to avoid network congestion. Existing methods often fail to accurately estimate key performance metrics, leading to suboptimal throughput and delayed data transfer. The invention provides a method for enhancing TCP performance by incorporating additional estimates into the congestion control process. These estimates include the number of round-trips required for TCP to reach a slow-start threshold and the bandwidth-delay-product (BDP), which represents the maximum data that can be in transit without causing congestion. By calculating these metrics, the system can better predict network conditions and adjust transmission rates more effectively. The method involves monitoring network traffic, analyzing packet round-trip times, and dynamically updating the estimates to reflect real-time conditions. This allows TCP to transition from slow-start to congestion avoidance more efficiently, reducing latency and improving overall throughput. The invention is particularly useful in high-latency or variable-bandwidth networks, where traditional congestion control algorithms struggle to maintain optimal performance.
11. The method of claim 9 , wherein the TCP estimates include a number of bytes transferred before TCP reaches a slow-start threshold or a bandwidth-delay-product.
A method for optimizing network communication involves estimating Transmission Control Protocol (TCP) performance metrics to improve data transfer efficiency. The technique addresses the challenge of inefficient data transmission in networked systems, particularly when TCP connections are initialized or encounter varying network conditions. The method calculates TCP estimates, including the number of bytes transferred before TCP reaches a slow-start threshold or a bandwidth-delay-product (BDP). These estimates help determine optimal data transfer rates and avoid congestion by dynamically adjusting transmission parameters. The slow-start threshold is a critical point in TCP's congestion control algorithm where the transmission rate transitions from exponential growth to linear growth. The BDP represents the product of available bandwidth and round-trip time, a key factor in maximizing throughput without overwhelming the network. By monitoring these metrics, the system can preemptively adjust transmission rates to maintain stable and efficient data flow. This approach enhances network performance by reducing latency and packet loss, particularly in environments with fluctuating bandwidth or high latency. The method integrates with existing TCP protocols to provide real-time adjustments, ensuring adaptive and responsive data transfer.
12. The method of claim 1 , wherein the first number is determined by: determining an underperformance value; estimating a slow start threshold; calculating an initial number of network round-trips that occur between a beginning of a response and a slow start threshold; calculating a subsequent number of network round-trips that occur between the slow start threshold until a fair-bandwidth-delay product is reached; determining a total number of network round-trips based on the initial number of network round-trips, the subsequent number of network round-trips, and the underperformance value; and computing the first number based on the total number of network round-trips and the bandwidth-delay product.
This invention relates to optimizing network communication performance, particularly in systems where initial data transmission rates are adjusted to improve efficiency. The problem addressed is the underperformance of network connections during the initial phase of data transfer, often due to conservative slow-start mechanisms that delay reaching optimal throughput. The method involves determining an underperformance value, which quantifies the inefficiency in the initial data transmission phase. A slow-start threshold is estimated, representing the point at which the transmission rate begins to increase more aggressively. The method calculates the number of network round-trips occurring between the start of the response and the slow-start threshold, as well as the subsequent round-trips needed to reach a fair-bandwidth-delay product, which represents the optimal transmission rate for the network path. These values are combined with the underperformance value to determine the total number of round-trips required to achieve efficient data transfer. Finally, the first number, which likely represents an initial transmission parameter, is computed based on the total round-trips and the bandwidth-delay product, ensuring faster convergence to optimal performance. This approach aims to minimize latency and maximize throughput during the initial phase of network communication.
13. The method of claim 1 , further comprising using an adaptive bit-rate algorithm to select a bitrate of outstanding video segments.
This invention relates to video streaming systems, specifically addressing the challenge of efficiently delivering video content over networks with varying bandwidth conditions. The method involves dynamically adjusting the bitrate of video segments to optimize streaming performance. An adaptive bit-rate algorithm is used to select the appropriate bitrate for outstanding video segments, which are segments of video data that have been requested but not yet transmitted. The algorithm evaluates network conditions, such as available bandwidth and latency, to determine the optimal bitrate for each segment. This ensures smooth playback by preventing buffering delays while maximizing video quality. The system may also include a buffer management component to monitor the buffer state of the client device and adjust the bitrate selection accordingly. Additionally, the method may involve pre-fetching video segments at different bitrates to allow quick switching in response to changing network conditions. The overall goal is to provide a seamless viewing experience by dynamically adapting to network fluctuations without interrupting playback.
14. An apparatus having improved video streaming performance, comprising: a transceiver; memory; and at least one processor operatively connected to the memory and the transceiver, the at least one processor being operative to: (i) determine a first number based on one or more parameters, at least one of the parameters being related to current network conditions; and (ii) at least one of: (a) determine a second number corresponding to a number of video segments of the video, as calculated by a total size of the video segments, that is greater than or equal in size to a third number determined based on at least a bandwidth-delay product of the network to the remote machine, the third number being no less than two, then request the second number of video segments in a pipelined fashion, and stop sending pipelined requests if a predetermined size of the video has been requested that is greater than or equal to the first number, and (b) request that the remote machine send a portion of the video, the portion of the video having a size that is equal to the first number or equal to the size of video remaining if less than the first number.
The invention relates to improving video streaming performance by dynamically adjusting the number of video segments requested based on network conditions. The apparatus includes a transceiver, memory, and at least one processor. The processor determines a first number based on network conditions, such as bandwidth and latency. It then either requests a second number of video segments in a pipelined manner, where the second number is calculated from the total size of the segments and must be at least two, or requests a portion of the video equal to the first number or the remaining video size if smaller. Pipelined requests stop when the requested video size reaches or exceeds the first number. This approach optimizes streaming efficiency by adapting to network constraints, reducing buffering delays, and ensuring smooth playback. The system dynamically balances between prefetching multiple segments and requesting specific portions to maintain performance under varying network conditions.
15. The apparatus of claim 14 , wherein the processor determines the first number based on two or more parameters, at least two of the parameters being related to network conditions.
This invention relates to a network optimization apparatus that dynamically adjusts operational parameters based on real-time network conditions. The apparatus includes a processor configured to determine a first number, which represents a key performance metric or control value, by analyzing two or more parameters. At least two of these parameters are related to network conditions, such as latency, bandwidth, packet loss, or signal strength. The processor uses these parameters to calculate the first number, which may then be applied to optimize network performance, improve resource allocation, or enhance communication efficiency. The apparatus may also include a memory for storing network data and a communication interface for receiving network condition inputs. The processor may further adjust the first number in response to changes in the network conditions, ensuring adaptive and efficient network operation. This invention addresses the challenge of maintaining optimal network performance under varying conditions by dynamically adjusting critical parameters based on real-time network metrics.
16. The apparatus of claim 14 , wherein the apparatus is a server, workstation, desktop computer, laptop, smart phone or mobile device, wearable device, smart TV, video-game console, digital video recorder, digital-media center, projector, tablet, set-top box, streaming stick, dongle, smart hub, or gateway.
This invention relates to a computing apparatus designed for processing and managing digital media content. The apparatus addresses the challenge of efficiently handling media data across various devices, ensuring compatibility, performance, and user accessibility. The core functionality involves receiving, processing, and transmitting digital media content, such as video, audio, or images, while optimizing resource usage and user experience. The apparatus includes a processor, memory, and communication interfaces to facilitate media processing tasks. It supports operations like decoding, encoding, transcoding, and streaming media content. The system may also include input/output interfaces for connecting to displays, storage devices, or network systems. Additionally, it may incorporate user interface components, such as touchscreens or remote controls, to enable interaction with the media content. The apparatus is adaptable to various form factors, including servers, workstations, desktop and laptop computers, smartphones, mobile devices, wearable devices, smart TVs, video-game consoles, digital video recorders, digital-media centers, projectors, tablets, set-top boxes, streaming sticks, dongles, smart hubs, and gateways. This versatility ensures the apparatus can be deployed in diverse environments, from home entertainment systems to enterprise-level media processing setups. The design prioritizes scalability, allowing integration with existing media ecosystems while maintaining high performance and reliability.
Unknown
September 8, 2020
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